1. Field of the Invention
The present invention relates in general to digital signal processing and in particular to circuits systems and methods for volume control in 1-bit digital audio systems.
2. Description of the Related Art
To date, most audio formats have used PCM encoding or an encoding that is subsequently decoded into a PCM format. As an example, a compact disk is recorded with a 16-bit format, and at a 44100 Hz sample rate. Newer audio formats use up to 24 bits, and up to 192 kHz sample rates. The Digital to Analog Converter (DAC) subsystem receives PCM data and passes those data through an interpolation filter to increase the sample rate. A delta-sigma modulator then reduces the number of bits representing each sample, for example from 24-bit samples to 1-bit samples (in a single bit modulator) or to 4-bit samples (in a multi-bit modulator). Modulator performance is typically specified in terms of its Modulation Index or Ml, which is the ratio of the maximum allowable signal peak modulator input to the mathematical maximal modulator input (equivalent to the signal peak of the feedback signal). For example, if the MI is 0.5 and the modulator has a single-bit bipolar output range, the maximum allowable input produces an output that is 75% +1 and 25% −1 for an average of 0.5. The delta-sigma modulator creates significant quantization noise; however, the delta-sigma modulator has the ability to shift this self-generated noise out of the signal band.
One advantage of multi-bit systems is that a higher modulation index can be used, meaning that the output signals can be of a greater level. The greater signal level directly improves the signal to noise level. If a multi bit modulator is used, the 4-bit data from the delta-sigma modulator is next thermometer encoded to represent 16 levels. The thermometer encoded data is passed through dynamic element matching logic implementing an algorithm for shaping the noise to account for digital to analog converter (DAC) element mismatch. The DAC, which ultimately converts the digital data to analog for eventual presentation to the listener as audio, is often a switched-capacitor circuit that also provides filtering, although continuous time circuits can also be used.
The newer Sony/Philips 1-bit recording system (“Super Audio CD” or “SADC”) stores data from an analog modulator onto the given digital storage media in a 1-bit format. As a result, techniques for converting data in the 1-bit digital format to analog must be developed. This is a non-trivial problem since such factors as filtering out of band noise, gain control through the modulator, and hardware minimization must be considered. Moreover, it is usually a requirement that a dynamic range of −120 dB in the audio band be achieved.
An additional consideration in 1-bit audio systems is volume control. Current 1-bit audio systems perform volume control operations in the analog domain, typically after the D/A conversion process. Analog volume controls are inherently undesirable because of additional power consumption, linearity issues, and the potential for added noise. Moreover, it is possible to convert the 1-bit data back into PCM data and then applying normal digital gain techniques. In this case, however, a decimation operation introducing a low pass function would be required which would add time effects that 1-bit encoding was designed to eliminate.
Given the potential for wide acceptance of the Sony/Philips 1-bit audio format, and the continuous demand for improved sound quality, circuits, systems and methods for volume control in 1-bit digital audio processing systems is required.